In order to digitally analyze the measuring results of sensors, it is necessary to convert the analog sensor signal or the signal sampled by sensors into a digital signal. For this purpose, analog/digital converters are frequently used which use the advantages of oversampling. In this case, the narrowband input signal is sampled at a high-frequency clock and subsequently digitized with the aid of an analog/digital converter. The bandwidth of the useful signal (also referred to as a useful band or baseband) is in this case significantly smaller than half of the sampling frequency. If the input signal includes high-frequency interference signals, these interference signals are folded down into the useful band, if necessary, due to the alias effect. In order to prevent this from happening, an anti-aliasing filter is usually used which filters out the high-frequency interference signals prior to the sampling.
FIG. 1 schematically shows a frequency diagram for such a situation. Analog useful signal 16 is sampled at a sampling frequency f. In an idealized circuit or an idealized method, this takes place for example, with the aid of a periodic clock signal having the period T=1/f, a sampling value being detected in each case at a certain point in time within the period, e.g., when the periodic clock signal exceeds or falls below a certain voltage, so that the time interval between two consecutive sampling points in time corresponds to period T in each case. The sampled useful signal is then obtained in the chronological representation as a product of the input signal and a sampling function which is defined by a sequence of equidistant sampling pulses in time interval T. In the frequency representation, this product corresponds to a folding of the frequency spectrum of the input signal with the frequency spectrum of the sampling function which is defined by a sequence of equidistant spectral lines at interval T. Frequency spectrum 10 of a sampling function is illustrated in FIG. 1. The component, which is centered around 0*f, of frequency spectrum 10 of the sampling function is not shown in the present case for the sake of a clearer illustration.
In a real circuit, inevitable stochastic deviations from this idealized behavior occur. For this reason, the sampling points in time and thus the sampling pulses in the sampling function are slightly shifted compared to the idealized sampling points in time, which is reflected in the frequency spectrum of sampling function 10 in that the idealized spectral lines are blurred across the frequency ranges, which are centered around the integral multiples of the sampling frequency f, and are increasingly attenuated as the frequency increases.
The input signal may include superimposed high-frequency interference signals such as those which occur during electromagnetic coupling. According to the related art, such an interference signal is filtered out using transfer function 14 prior to the sampling with the aid of an anti-aliasing filter, e.g., a low-pass filter. Such a filtered, high-frequency interference signal 12 is shown, as an example, in FIG. 1 at a frequency which is slightly above twice the sampling frequency f. The point of intersection between transfer function 14 of the anti-aliasing filter and the frequency axis delimits baseband 1 to a range which extends from 0*f up to a frequency of f/2. The low-pass filter is in this case designed in such a way that an analog useful signal 16 may pass preferably unfiltered in baseband 1 at a typical useful frequency. The disadvantage of this approach is that an anti-aliasing filter must be implemented. In particular an implementation on an application-specific integrated circuit (ASIC) results in development complexity and space consumption.
FIG. 2 shows the frequency spectra of the sampling of an analog useful signal 16 without an anti-aliasing filter. Baseband 1 extends across a frequency range of 0*f through f/2 as in the case of filtering using an anti-aliasing filter. However, interference amplitude 20, which represents the folding of an unfiltered, high-frequency interferer 18 in the case of sampling without using the anti-aliasing filter, is folded directly into baseband 1 in most cases and therefore wrongly interpreted as a useful signal 16 during sampling at a sampling frequency f, frequency spectrum 10 of which is plotted in FIG. 2 identically to that in FIG. 1.
A method for sampling an analog input signal and for storing a digital representation of the sampling values in a memory is known from DE 602 12 795 T2, a sequence of clock pulses being generated at a predefined frequency Fclk and being alternated with the aid of the generated pseudo random sequences. The pseudo-randomly alternated clock pulses are used for creating a sequence of sampling pulses.
The subject matter described in DE 602 12 795 T2 is based on a teaching which is described in I. Bilinskis, G. Cain, Digital Alias-Free Signal Processing in the GHz Frequency Range, XP-001133893, Institution of Electrical Engineers, 1996, p. 6/1-6/6.
Furthermore, a method for analog/digital conversion and digital signal analysis as well as a device for carrying out the method is known from DE 24 55 302 A1, the analog input signal being subjected to a sampling from equidistant and (at least pseudo) stochastically varied sampling intervals which are superposed on one another.